Chapter 3 Finalversion

Chapter 3
Transport
Layer
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Computer Networking: A
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Top-Down Approach
Thanks and enjoy! JFK/KWR 8th edition
Jim Kurose, Keith Ross
All material copyright 1996-2020
J.F Kurose and K.W. Ross, All Rights Reserved Pearson, 2020
Transport Layer: 3-1
Transport layer: roadmap
 Transport-layer services
 Multiplexing and demultiplexing
 Connectionless transport: UDP
 Principles of reliable data transfer
 Connection-oriented transport: TCP
 Principles of congestion control
 TCP congestion control

Transport services and protocols
application
transport
 provide logical communication mobile

network
network
data link
physical
between application processes national or global ISP
running on different hosts
log
ica
le
 transport protocols actions in end
nd
-e
systems:
nd
local or
tra
• sender: breaks application messages regional ISP
n sp
into segments, passes to network layer
ort
home network content
• receiver: reassembles segments into provider
network
messages, passes to application layer application
transport
datacenter
network
network
 two transport protocols available to data link

physical
Internet applications enterprise

network
• TCP, UDP
Transport vs. network layer services and protocols
household analogy:
12 kids in Ann’s house sending
letters to 12 kids in Bill’s house:
 hosts = houses
 processes = kids
 app messages = letters in
envelopes
 transport protocol = Ann and Bill
who demux to in-house siblings
 network-layer protocol = postal
service

Transport vs. network layer services and protocols
 network layer: logical household analogy:

communication between 12 kids in Ann’s house sending
hosts letters to 12 kids in Bill’s house:
 hosts = houses
 transport layer: logical  processes = kids
communication between  app messages = letters in
processes envelopes
• relies on, enhances, network  transport protocol = Ann and Bill
layer services who demux to in-house siblings
 network-layer protocol = postal
service

Transport Layer Actions
Sender:
application  is passed an application- app. msg
application
layer message
transport
 determines segment TTh htransport
app. msg
header fields values
network (IP)
 creates segment network (IP)
link
 passes segment to IP link
physical physical

Transport Layer Actions
Receiver:
application  receives segment from IP application
 checks header values
app. msg
transport  extracts application-layer transport
message
network (IP)  demultiplexes message up network (IP)
link to application via socket link
physical physical
Th app. msg

Two principal Internet transport protocols
application
transport
 TCP: Transmission Control Protocol mobile

network
network
data link
physical
national or global ISP
• reliable, in-order delivery
log
• congestion control
ica
le
• flow control
nd
-e
• connection setup
nd
local or
tra
regional ISP
 UDP: User Datagram Protocol
n sp
ort
• unreliable, unordered delivery home network content
provider
• no-frills extension of “best-effort” IP network
application
transport
datacenter
network
 services not available:

network
data link
physical
• delay guarantees enterprise

network
• bandwidth guarantees
Chapter 3: roadmap
 Evolution of transport-layer
functionality
Why Multiplexing & demultiplexing……
application application
HTTP msg
transport
transport network transport

network link network
link physical link
physical physical
client
HTTP server Transport Layer: 3-10
HTTP server
client
HTTP msg
Ht HTTP msg
transport

link physical link
physical physical

HTTP server
client
HTTP msg
Ht HTTP msg
transport
Hnnetwork
Ht HTTP msg
transport transport
link physical link
physical physical

HTTP server
client
transport

link physical link
physical physical
Hn Ht HTTP msg

HTTP server
client1 client2
application P-client1 P-client2 application
transport

link physical link
physical physical

Multiplexing/demultiplexing
multiplexing at sender: demultiplexing at receiver:
handle data from multiple use header info to deliver
sockets, add transport header received segments to correct
(later used for demultiplexing) socket
application
application P1 P2 application socket

P3 transport P4
process
link physical link
physical physical

How demultiplexing works
 host receives IP datagrams 32 bits
• each datagram has source IP source port # dest port #
address, destination IP address
• each datagram carries one other header fields
transport-layer segment
• each segment has source,
application
destination port number data
 host uses IP addresses & port (payload)
numbers to direct segment to
appropriate socket TCP/UDP segment format

Connectionless demultiplexing
Recall: when receiving host receives
 when creating socket, must UDP segment:
• checks destination port # in
specify host-local port #:
segment
DatagramSocket mySocket1 = new
DatagramSocket(12534); • directs UDP segment to socket
with that port #
 when creating datagram to
send into UDP socket, must
IP/UDP datagrams with same dest.
specify port #, but different source IP
• destination IP address addresses and/or source port
• destination port # numbers will be directed to same
socket at receiving host
Connectionless demultiplexing: an example
DatagramSocket
serverSocket = new
DatagramSocket
DatagramSocket mySocket2 = DatagramSocket mySocket1 =
new DatagramSocket (6428); new DatagramSocket (5775);
(9157); application
application P1 application
P3 P4
transport
transport transport
network
link physical link
physical physical
source port: 6428 source port: ?

dest port: 9157 dest port: ?
source port: 9157 source port: ?

dest port: 6428 dest port: ?
Connection-oriented demultiplexing
 TCP socket identified by  server may support many
4-tuple: simultaneous TCP sockets:
• source IP address • each socket identified by its
• source port number own 4-tuple
• dest IP address • each socket associated with
• dest port number a different connecting client
 demux: receiver uses all
four values (4-tuple) to
direct segment to
appropriate socket
Connection-oriented demultiplexing: example
application
application P4 P5 P6 application
P1 P2 P3
transport
transport transport
network
link physical link
physical server: IP physical
address B
host: IP source IP,port: B,80 host: IP

address A dest IP,port: A,9157 source IP,port: C,5775 address C
dest IP,port: B,80
source IP,port: A,9157
dest IP, port: B,80
source IP,port: C,9157
dest IP,port: B,80
Three segments, all destined to IP address: B,
dest port: 80 are demultiplexed to different sockets
Summary
 Multiplexing, demultiplexing: based on segment, datagram
header field values
 UDP: demultiplexing using destination port number (only)
 TCP: demultiplexing using 4-tuple: source and destination IP
addresses, and port numbers
 Multiplexing/demultiplexing happen at all layers

Chapter 3: roadmap
functionality
UDP: User Datagram Protocol
Why is there a UDP?
 “no frills,” “bare bones”  no connection
Internet transport protocol establishment (which can
add RTT delay)
 “best effort” service, UDP  simple: no connection state
segments may be: at sender, receiver [Do not
• lost handle Buffer size, congestion control
parameters, Seq and Ack numbers]
• delivered out-of-order to app
 small header size
 connectionless:  no congestion control
• no handshaking between UDP  UDP can blast away as fast as
sender, receiver desired!
• each UDP segment handled  can function in the face of
independently of others congestion
UDP: User Datagram Protocol
 UDP use:
 streaming multimedia apps (loss tolerant, rate sensitive)
 DNS
 SNMP
 HTTP/3
 if reliable transfer needed over UDP (e.g., HTTP/3):
 add needed reliability at application layer
 add congestion control at application layer

UDP: User Datagram Protocol [RFC 768]

UDP segment header
32 bits
source port # dest port #
length checksum
application length, in bytes of

data UDP segment,
(payload) including header
data to/from
UDP segment format application layer

UDP checksum
Goal: detect errors (i.e., flipped bits) in transmitted segment
1st number 2nd number sum
Transmitted: 5 6 11
Received: 4 6 11
receiver-computed
checksum
= sender-computed
checksum (as received)

UDP checksum
Goal: detect errors (i.e., flipped bits) in transmitted segment
sender: receiver:
 treat contents of UDP  compute checksum of received
segment (including UDP header segment
fields and IP addresses) as
sequence of 16-bit integers  check the checksum value
 checksum: addition (one’s calculated by receiver:
complement sum) of segment • checksum of receiver  all zero bits -
content no error
 checksum value put into • checksum of receiver  any bit non-
zero – error present
UDP checksum field
Internet checksum: an example
example: add two 16-bit integers
1110011001100110
1101010101010101
wraparound 11011101110111011
sum 1011101110111100
checksum 0100010001000011
Note: when adding numbers, a carryout from the most significant bit needs to be
added to the result
* Check out the online interactive exercises for more examples: h ttp://gaia.cs.umass.edu/kurose_ross/interactive/
Internet checksum: weak protection!
example: add two 16-bit integers
01
1110011001100110 10
1101010101010101
wraparound 11011101110111011 Even though
numbers have
sum 1011101110111100 changed (bit
flips), no change
checksum 0100010001000011 in checksum!

Summary: UDP
 “no frills” protocol:
• segments may be lost, delivered out of order
• best effort service: “send and hope for the best”
 UDP has its plusses:
• no setup/handshaking needed (no RTT incurred)
• can function when network service is compromised
• helps with reliability (checksum)
 build additional functionality on top of UDP in application layer
(e.g., HTTP/3)
Chapter 3: roadmap
functionality
Principles of reliable data transfer
sending receiving
process process
application data data
transport
reliable channel
reliable service abstraction

sending receiving sending receiving

process process process process
application data data application data data
transport transport
reliable channel
sender-side of receiver-side
reliable service abstraction reliable data of reliable data
transfer protocol transfer protocol
transport
network
unreliable channel
reliable service implementation

sending receiving
process process
transport
Complexity of reliable data reliable data
transfer protocol
of reliable data
transfer protocol
transfer protocol will depend
(strongly) on characteristics of transport
network
unreliable channel (lose, unreliable channel
corrupt, reorder data?)

sending receiving
process process
transport
reliable data of reliable data
Sender, receiver do not know transfer protocol transfer protocol
the “state” of each other, e.g.,
was a message received? transport
network
 unless communicated via a unreliable channel
message

Reliable data transfer protocol (rdt): interfaces
rdt_send(): called from above, deliver_data(): called by rdt to
(e.g., by app.). Passed data to deliver data to upper layer
deliver to receiver upper layer
sending receiving
process process
rdt_send() data data
deliver_data()
sender-side data receiver-side

implementation of implementation of
rdt reliable data packet rdt reliable data
transfer protocol transfer protocol
udt_send() Header data Header data rdt_rcv()
unreliable channel
udt_send(): called by rdt rdt_rcv(): called when packet
to transfer packet over Bi-directional communication over arrives on receiver side of
unreliable channel to receiver unreliable channel channel
Reliable data transfer: getting started
We will:
 incrementally develop sender, receiver sides of reliable data transfer
protocol (rdt)
 consider only unidirectional data transfer
• but control info will flow in both directions!
 use finite state machines (FSM) to specify sender, receiver
event causing state transition
actions taken on state transition
state: when in this “state”
next state uniquely state state
determined by next 1 event
event 2
actions

rdt1.0: reliable transfer over a reliable channel
 underlying channel perfectly reliable
• no bit errors
• no loss of packets
 separate FSMs for sender, receiver:

• sender sends data into underlying channel
• receiver reads data from underlying channel
Wait for rdt_send(data) Wait for rdt_rcv(packet)

sender call from packet = make_pkt(data) receiver call from extract (packet,data)
above udt_send(packet) below deliver_data(data)

rdt2.0: channel with bit errors
 underlying channel may flip bits in packet
• checksum (e.g., Internet checksum) to detect bit errors
 the question: how to recover from errors?
How do humans recover from “errors” during conversation?

rdt2.0: channel with bit errors
 underlying channel may flip bits in packet
• checksum to detect bit errors
 the question: how to recover from errors?
• acknowledgements (ACKs): receiver explicitly tells sender that pkt
received OK
• negative acknowledgements (NAKs): receiver explicitly tells sender
that pkt had errors
• sender retransmits pkt on receipt of NAK
stop and wait

sender sends one packet, then waits for receiver response
rdt2.0: FSM specifications
rdt_send(data)
snkpkt = make_pkt(data, checksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
Wait for Wait for isNAK(rcvpkt)
sender call from ACK or udt_send(sndpkt) rdt_rcv(rcvpkt) && corrupt(rcvpkt)
above NAK
udt_send(NAK)
rdt_rcv(rcvpkt) && isACK(rcvpkt)

Wait for
L
call from receiver
below
rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)

extract(rcvpkt,data)
deliver_data(data)
udt_send(ACK)

rdt2.0: FSM specification
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
above NAK
udt_send(NAK)

Wait for
L
call from receiver
below
Note: “state” of receiver (did the receiver get my

message correctly?) isn’t known to sender unless
somehow communicated from receiver to sender deliver_data(data)
 that’s why we need a protocol! udt_send(ACK)

rdt2.0: operation with no errors
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
above NAK
udt_send(NAK)

Wait for
L call from receiver
below

deliver_data(data)
udt_send(ACK)

rdt2.0: corrupted packet scenario
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
above NAK
udt_send(NAK)

Wait for
L call from receiver
below

deliver_data(data)
udt_send(ACK)

rdt2.0 has a fatal flaw!
what happens if ACK/NAK handling duplicates:
corrupted?  sender retransmits current pkt if
 sender doesn’t know what ACK/NAK corrupted
happened at receiver!  sender adds sequence number to
 can’t just retransmit: possible each pkt
duplicate
 receiver discards (doesn’t deliver
up) duplicate pkt
stop and wait

sender sends one packet, then
waits for receiver response
(EXTRA for interest)
rdt2.1: sender, handling garbled ACK/NAKs
rdt_send(data)
sndpkt = make_pkt(0, data, checksum)
udt_send(sndpkt) rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) ||
Wait for Wait for isNAK(rcvpkt) )
call 0 from ACK or
NAK 0 udt_send(sndpkt)
above
rdt_rcv(rcvpkt)
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) &&
&& notcorrupt(rcvpkt)
isACK(rcvpkt)
&& isACK(rcvpkt)
L
L
Wait for Wait for
ACK or call 1 from
rdt_rcv(rcvpkt) NAK 1 above
&& (corrupt(rcvpkt) ||
isNAK(rcvpkt) ) rdt_send(data)
udt_send(sndpkt) sndpkt = make_pkt(1, data, checksum)

udt_send(sndpkt)

rdt2.1: receiver, handling garbled ACK/NAKs
&& has_seq0(rcvpkt)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)
rdt_rcv(rcvpkt) && rdt_rcv(rcvpkt) &&
(corrupt(rcvpkt) (corrupt(rcvpkt)
sndpkt = make_pkt(NAK, chksum) sndpkt = make_pkt(NAK, chksum)
udt_send(sndpkt) udt_send(sndpkt)
Wait for Wait for
rdt_rcv(rcvpkt) && 0 from 1 from rdt_rcv(rcvpkt) &&
not corrupt(rcvpkt) && below below not corrupt(rcvpkt) &&
has_seq1(rcvpkt) has_seq0(rcvpkt)
sndpkt = make_pkt(ACK, chksum) sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt) udt_send(sndpkt)
&& has_seq1(rcvpkt)
deliver_data(data)
sndpkt = make_pkt(ACK, chksum)
udt_send(sndpkt)

rdt2.1: discussion
sender: receiver:
 seq # added to pkt  must check if received packet
 two seq. #s (0,1) will suffice. is duplicate
Why? • state indicates whether 0 or 1 is
expected pkt seq #
 must check if received ACK/NAK
 note: receiver can not know if
corrupted
its last ACK/NAK received OK
 twice as many states at sender
• state must “remember” whether
“expected” pkt should have seq #
of 0 or 1

rdt2.2: a NAK-free protocol
 same functionality as rdt2.1, using ACKs only
 instead of NAK, receiver sends ACK for last pkt received OK
• receiver must explicitly include seq # of pkt being ACKed
 duplicate ACK at sender results in same action as NAK:
retransmit current pkt
As we will see, TCP uses this approach to be NAK-free

rdt2.2: sender, receiver fragments
rdt_send(data)
udt_send(sndpkt)
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) ||
Wait for Wait for
ACK isACK(rcvpkt,1) )
call 0 from
above 0 udt_send(sndpkt)
sender FSM
fragment rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt)
rdt_rcv(rcvpkt) && && isACK(rcvpkt,0)
(corrupt(rcvpkt) || L
has_seq1(rcvpkt)) Wait for receiver FSM
0 from
udt_send(sndpkt) below fragment
&& has_seq1(rcvpkt)
deliver_data(data)
sndpkt = make_pkt(ACK1, chksum)
udt_send(sndpkt) Transport Layer: 3-51
rdt3.0: channels with errors and loss
New channel assumption: underlying channel can also lose
packets (data, ACKs)
• checksum, sequence #s, ACKs, retransmissions will be of help …
but not quite enough
Q: How do humans handle lost sender-to-

receiver words in conversation?

rdt3.0: channels with errors and loss
Approach: sender waits “reasonable” amount of time for ACK
 retransmits if no ACK received in this time
 if pkt (or ACK) just delayed (not lost):
• retransmission will be duplicate, but seq #s already handles this!
• receiver must specify seq # of packet being ACKed
 use countdown timer to interrupt after “reasonable” amount of
time
timeout

rdt3.0 sender
rdt_send(data)
udt_send(sndpkt)
start_timer
Wait for Wait

call 0 from for
above ACK0
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) rdt_rcv(rcvpkt)
&& isACK(rcvpkt,1) && notcorrupt(rcvpkt)
stop_timer && isACK(rcvpkt,0)
stop_timer
Wait Wait for
for call 1 from
ACK1 above
rdt_send(data)
udt_send(sndpkt)
start_timer

rdt3.0 sender
rdt_send(data)
rdt_rcv(rcvpkt) &&
sndpkt = make_pkt(0, data, checksum) ( corrupt(rcvpkt) ||
udt_send(sndpkt) isACK(rcvpkt,1) )
rdt_rcv(rcvpkt) start_timer L
L Wait for Wait
for timeout
call 0 from
ACK0 udt_send(sndpkt)
above
start_timer
rdt_rcv(rcvpkt)
&& notcorrupt(rcvpkt) rdt_rcv(rcvpkt)
&& isACK(rcvpkt,1) && notcorrupt(rcvpkt)
stop_timer && isACK(rcvpkt,0)
stop_timer
Wait Wait for
timeout for call 1 from
udt_send(sndpkt) ACK1 above
start_timer rdt_rcv(rcvpkt)
rdt_send(data) L
rdt_rcv(rcvpkt) &&
( corrupt(rcvpkt) || udt_send(sndpkt)
isACK(rcvpkt,0) ) start_timer
L

rdt3.0 in action
sender receiver sender receiver
send pkt0 pkt0 send pkt0 pkt0
rcv pkt0 rcv pkt0
ack0 send ack0 ack0 send ack0
rcv ack0 rcv ack0
send pkt1 pkt1 send pkt1 pkt1
rcv pkt1 X
loss
ack1 send ack1
rcv ack1
send pkt0 pkt0
rcv pkt0 timeout
ack0 send ack0 resend pkt1 pkt1
rcv pkt1
ack1 send ack1
rcv ack1
send pkt0 pkt0
(a) no loss rcv pkt0
ack0 send ack0
(b) packet loss

rdt3.0 in action
sender receiver
sender receiver send pkt0
pkt0
rcv pkt0
send pkt0 pkt0 send ack0
ack0
rcv pkt0 rcv ack0
ack0 send ack0 send pkt1 pkt1
rcv ack0 rcv pkt1
rcv pkt1 ack1
ack1 send ack1
X timeout
loss resend pkt1
pkt1 rcv pkt1
timeout
resend pkt1 pkt1
rcv pkt1 rcv ack1 (detect duplicate)
(detect duplicate)
ack1 send ack1 ack1 rcv pkt0
rcv ack1 rcv ack1 send ack0
send pkt0 pkt0 (ignore) ack0
rcv pkt0
ack0 send ack0 pkt1
(c) ACK loss (d) premature timeout/ delayed ACK

rdt3.0: stop-and-wait operation
sender receiver
L/R L/R
Usender =
RTT + L / R
.008 RTT
=
30.008
= 0.00027
 rdt 3.0 protocol performance stinks!

 Protocol limits performance of underlying infrastructure (channel)

rdt3.0: pipelined protocols operation
pipelining: sender allows multiple, “in-flight”, yet-to-be-acknowledged
packets
• range of sequence numbers must be increased
• buffering at sender and/or receiver

Pipelining: increased utilization
sender receiver
first packet bit transmitted, t = 0
last bit transmitted, t = L / R
first packet bit arrives

RTT last packet bit arrives, send ACK
last bit of 2nd packet arrives, send ACK
last bit of 3rd packet arrives, send ACK
ACK arrives, send next
packet, t = RTT + L / R
3-packet pipelining increases
utilization by a factor of 3!

Go-Back-N: sender
 sender: “window” of up to N, consecutive transmitted but unACKed pkts
• k-bit seq # in pkt header
 cumulative ACK: ACK(n): ACKs all packets up to, including seq # n

• on receiving ACK(n): move window forward to begin at n+1
 timer for oldest in-flight packet
 timeout(n): retransmit packet n and all higher seq # packets in window
Go-Back-N: receiver
 ACK-only: always send ACK for correctly-received packet so far, with
highest in-order seq #
• may generate duplicate ACKs
• need only remember rcv_base
 on receipt of out-of-order packet:
• can discard (don’t buffer) or buffer: an implementation decision
• re-ACK pkt with highest in-order seq #
Receiver view of sequence number space:

received and ACKed
… … Out-of-order: received but not ACKed
rcv_base
Not received
Go-Back-N in action
sender window (N=4) sender receiver
012345678 send pkt0
012345678 send pkt1
send pkt2 receive pkt0, send ack0
012345678
send pkt3 Xloss receive pkt1, send ack1
012345678
(wait)
receive pkt3, discard,
012345678 rcv ack0, send pkt4 (re)send ack1
012345678 rcv ack1, send pkt5 receive pkt4, discard,
(re)send ack1
ignore duplicate ACK receive pkt5, discard,
(re)send ack1
pkt 2 timeout
012345678 send pkt2
012345678 send pkt3
012345678 send pkt4 rcv pkt2, deliver, send ack2
012345678 send pkt5 rcv pkt3, deliver, send ack3
rcv pkt4, deliver, send ack4
rcv pkt5, deliver, send ack5

Selective repeat
 A single packet error can cause GBN to retransmit a large
number of packets when the packet size is very large.
 Receiver individually acknowledges all correctly received packets
• Buffers packets, as needed, for eventual in-order delivery to the upper
layer
 Sender times-out/retransmits individually for unACKed packets
• Sender maintains a timer for each unACKed pkt
 sender window
• N consecutive seq #s
• limits seq #s of sent, unACKed packets

Selective repeat: sender, receiver windows

Selective Repeat in action
sender window (N=4) sender receiver
012345678 send pkt0
012345678 send pkt1
012345678 send pkt2 receive pkt0, send ack0
012345678 send pkt3 Xloss receive pkt1, send ack1
(wait)
receive pkt3, buffer,
012345678 rcv ack0, send pkt4 send ack3
012345678 rcv ack1, send pkt5
record ack3 arrived send ack4
pkt 2 timeout send ack5
012345678 send pkt2
012345678 (but not 3,4,5)
012345678 rcv pkt2; deliver pkt2,
012345678 pkt3, pkt4, pkt5; send ack2
Q: what happens when ack2 arrives?

sender window receiver window
(after receipt) (after receipt)
Selective repeat: dilemma 0123012 pkt0
0123012 pkt1 0123012
0123012 pkt2 0123012
example: 0123012
0123012 pkt3
 seq #’s: 0, 1, 2, 3 0123012
X
pkt0
 window size=3 will accept packet
with seq number 0
(a) no problem
 receiver sees no difference in two
scenarios! receiver can’t see sender side.
 duplicate data accepted as new in receiver behavior identical in both cases!
something’s (very) wrong!
(b)
0123012 pkt0
Q: what relationship between seq # 0123012 pkt1 0123012

pkt2
size and window size to avoid 0123012
X
0123012
0123012
problem in (b)? X
timeout
retransmit pkt0 X
0123012 pkt0
will accept packet
with seq number 0
less than or equal to the half of the sequence (b) oops!
number 3-67
Chapter 3: roadmap
• segment structure
• reliable data transfer
• flow control
• connection management
TCP: overview RFCs: 793,1122, 2018, 5681, 7323
 point-to-point:  cumulative ACKs
• one sender, one receiver
 pipelining:
 reliable, in-order byte stream: • TCP congestion and flow control
• no “message boundaries" set window size
 full duplex data:  connection-oriented:
• bi-directional data flow in same
connection • handshaking (exchange of control
• MSS: maximum segment size messages) initializes sender,
MSS: maximum amount of receiver state before data exchange
application layer data in the  flow controlled:
segment. • sender will not overwhelm receiver

TCP Seq numbers and Acks
 Suppose Host A wants to send a stream of data to a process in Host B over
a TCP connection. Assume that, the data stream consists of a file consisting
of 500,000 bytes, and that the MSS is 1,000 bytes. Then the segment looks
like this:

TCP segment structure
32 bits
source port # dest port # segment seq #: counting

ACK: seq # of next expected sequence number bytes of data into bytestream
byte; A bit: this is an ACK (not segments!)
acknowledgement number
length (of TCP header) head not
len used C EUAP R SF receive window flow control: # bytes
Internet checksum checksum Urg data pointer receiver willing to accept
options (variable
C, E: congestion notification length)
TCP options
application data sent by
RST, SYN, FIN: connection data application into
management (variable length) TCP socket

TCP sequence numbers, ACKs
Sequence numbers: outgoing segment from sender
• byte stream “number” of first

source port # dest port #
sequence number
byte in segment’s data acknowledgement number
rwnd
Acknowledgements: checksum urg pointer
• seq # of next byte expected from window size

N
other side
• cumulative ACK
sender sequence number space
Q: how the receiver handles out-of-
order segments sent
ACKed
sent, not- usable not
yet ACKed but not usable
yet sent
• A: Discards out-of-order segments (“in-flight”)
• B: Keeps the out-of-order bytes in the outgoing segment from receiver

buffer source port # dest port #
sequence number
• C: TCP spec doesn’t say, - up to the acknowledgement number
implementor A
checksum
rwnd
urg pointer
TCP sequence numbers, ACKs (Some Scenarios)
 Host A has received all data from 0 to 535 and Host A is expecting data
536 and all subsequent byte streams from B.
 Host A received one segment from 0 through 535 and another segment
from 900 to 1000.
• It has not received any segment from 536 to 899. Therefore, there has a gap
• Cumulative acknowledgments.
 Host A received the segment from 900 to 1000 before receiving bytes
536 to 899. Therefore, out of order.

TCP sequence numbers, ACKs
Host A Host B
User types‘C’
Seq=42, ACK=79, data = ‘C’
host ACKs receipt of‘C’,
echoes back ‘C’
Seq=79, ACK=43, data = ‘C’
host ACKs receipt
of echoed ‘C’
Seq=43, ACK=80
simple telnet scenario

TCP round trip time, timeout
Q: how to set TCP timeout Q: how to estimate RTT?
value?  SampleRTT:measured time
 longer than RTT, but RTT varies! from segment transmission until
ACK receipt
 too short: premature timeout,
• ignore retransmissions
unnecessary retransmissions
 SampleRTT will vary, want
 too long: slow reaction to estimated RTT “smoother”
segment loss • average several recent
measurements, not just current
SampleRTT
SampleRTT values will fluctuate from segment to segment due to congestion
and load on the end systems.
EstimatedRTT = (1- )*EstimatedRTT + *SampleRTT
 exponential weighted moving average (EWMA)
 influence of past sample decreases exponentially fast
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
 typical value:  = 0.125 350
RTT: gaia.cs.umass.edu to fantasia.eurecom.fr
RTT (milliseconds)
300
250
RTT (milliseconds)
200
sampleRTT
150
EstimatedRTT
100
1 8 15 22 29 36 43 50 57 64 71 78 85 92 99 106
time (seconnds)
time (seconds)
SampleRTT Estimated RTT
 DevRTT: EWMA of SampleRTT deviation from EstimatedRTT:
DevRTT = (1-)*DevRTT + *|SampleRTT-EstimatedRTT|
(typically,  = 0.25)
 timeout interval: EstimatedRTT plus “safety margin”

• large variation in EstimatedRTT: want a larger safety margin
TimeoutInterval = EstimatedRTT + 4*DevRTT
estimated RTT “safety margin”
TCP: retransmission scenarios
Host A Host B Host A Host B
SendBase=92
Seq=92, 8 bytes of data Seq=92, 8 bytes of data
Seq=100, 20 bytes of data

timeout
timeout
ACK=100
X
ACK=100
ACK=120
Seq=92, 8 bytes of data Seq=92, 8

SendBase=100 bytes of data send cumulative
SendBase=120 ACK for 120
ACK=100
ACK=120
SendBase=120
lost ACK scenario premature timeout

TCP: retransmission scenarios
Host A Host B

ACK=100
X
ACK=120
cumulative ACK covers

for earlier lost ACK

TCP fast retransmit
Host A Host B
TCP fast retransmit
if sender receives 3 additional
Seq=92
ACKs for same data (“triple Seq=1
, 8 bytes
of data
duplicate ACKs”), resend unACKed 0 0, 20 b
ytes o
f data
segment with smallest seq # X
 likely that unACKed segment lost,
=100
so don’t wait for timeout ACK
=100
timeout
ACK
CK =100
A
=100
ACK
Receipt of three duplicate ACKs
indicates 3 segments received
after a missing segment – lost
segment is likely. So retransmit!
Chapter 3: roadmap
• segment structure
• reliable data transfer
• flow control
• connection management
TCP flow control
application
Q: What happens if network Application removing
process
layer delivers data faster than data from TCP socket

buffers
application layer removes TCP socket
data from socket buffers? receiver buffers
TCP
code
Network layer
delivering IP datagram
payload into TCP
IP
socket buffers code
from sender
receiver protocol stack

TCP flow control
application
process

buffers
TCP
code
Network layer
delivering IP datagram
payload into TCP
IP
socket buffers code
from sender

TCP flow control
application
process

buffers
TCP
code
receive window
flow control: # bytes
receiver willing to accept IP
code
from sender

TCP flow control
application
process

buffers
TCP
flow control code
receiver controls sender, so

sender won’t overflow IP
code
receiver’s buffer by
transmitting too much, too fast
from sender

TCP flow control
 TCP receiver “advertises” free buffer
space in rwnd field in TCP header to application process
• RcvBuffer size set via socket
options (typical default is 4096 bytes) RcvBuffer buffered data
• many operating systems autoadjust
rwnd free buffer space
RcvBuffer
 sender limits amount of unACKed
(“in-flight”) data to received rwnd TCP segment payloads
 guarantees receive buffer will not TCP receiver-side buffering

overflow
LastByteSent – LastByteAcked <= rwnd

TCP flow control
 LastByteRead: data stream read from the buffer
 LastByteRcvd: the data stream that has arrived
from the network
 LastByteRcvd – LastByteRead <= RcvBuffer
A
 Initially rwnd = RcvBuffer
A
 Host A to continue to send segments with one
data byte when B’s receive window is zero. These
segments will be acknowledged by the receiver
3-87
TCP connection management
before exchanging data, sender/receiver “handshake”:
 agree to establish connection (each knowing the other willing to establish connection)
 agree on connection parameters (e.g., starting seq #s)
connection state: ESTAB connection state: ESTAB

connection variables: connection Variables:
seq # client-to-server seq # client-to-server
server-to-client server-to-client
rcvBuffer size rcvBuffer size
at server,client at server,client
network network
Socket clientSocket = Socket connectionSocket =

newSocket("hostname","port number"); welcomeSocket.accept();
TCP 3-way handshake
Server state
serverSocket = socket(AF_INET,SOCK_STREAM)
Client state serverSocket.bind((‘’,serverPort))
serverSocket.listen(1)
clientSocket = socket(AF_INET, SOCK_STREAM) connectionSocket, addr = serverSocket.accept()
LISTEN
clientSocket.connect((serverName,serverPort)) LISTEN
choose init seq num, x
send TCP SYN msg
SYNSENT SYNbit=1, Seq=x
choose init seq num, y
send TCP SYNACK
msg, acking SYN SYN RCVD
SYNbit=1, Seq=y
ACKbit=1; ACKnum=x+1
received SYNACK(x)
ESTAB indicates server is live;
send ACK for SYNACK;
this segment may contain ACKbit=1, ACKnum=y+1
client-to-server data
received ACK(y)
indicates client is live
ESTAB

A human 3-way handshake protocol
1. On belay?
2. Belay on.
3. Climbing.

TCP 3-way handshake:
FSM
closed
Socket connectionSocket =
welcomeSocket.accept();
L Socket clientSocket =
SYN(x) newSocket("hostname","port
number");
SYNACK(seq=y,ACKnum=x+1)
create new socket for listen SYN(seq=x)
communication back to client
SYN SYN
rcvd sent
SYNACK(seq=y,ACKnum=x+1)
ESTAB ACK(ACKnum=y+1)
ACK(ACKnum=y+1)
L
Transport Layer 3-91

Closing a TCP connection
 client, server each close their side of connection
• send TCP segment with FIN bit = 1
 respond to received FIN with ACK
• on receiving FIN, ACK can be combined with own FIN
 simultaneous FIN exchanges can be handled

TCP: closing a connection
client state server state
ESTAB ESTAB
clientSocket.close()
FIN_WAIT_1 can no longer FINbit=1, seq=x
send but can
receive data CLOSE_WAIT
ACKbit=1; ACKnum=x+1
can still
FIN_WAIT_2 wait for server send data
close
LAST_ACK
FINbit=1, seq=y
TIMED_WAIT can no longer
send data
ACKbit=1; ACKnum=y+1
timed wait
for 2*max CLOSED
segment lifetime
CLOSED

TCP: closing a connection
Fig: A typical sequence of TCP states visited

by a client TCP
Fig: A typical sequence of TCP states visited

by a server TCP

Chapter 3: roadmap
functionality
TCP congestion control: AIMD
 approach: senders can increase sending rate until packet loss
(congestion) occurs, then decrease sending rate on loss event
Additive Increase Multiplicative Decrease
increase sending rate by 1 cut sending rate in half at
maximum segment size every each loss event
RTT until loss detected
TCP sender Sending rate
AIMD sawtooth
behavior: probing
for bandwidth
time Transport Layer: 3-96

TCP AIMD: more
Multiplicative decrease detail: sending rate is
 Cut in half on loss detected by triple duplicate ACK (TCP Reno)
 Cut to 1 MSS (maximum segment size) when loss detected by
timeout (TCP Tahoe)
Why AIMD?
 AIMD – a distributed, asynchronous algorithm – has been
shown to:
• optimize congested flow rates network wide!
• have desirable stability properties

TCP congestion control: details
sender sequence number space
cwnd TCP sending behavior:
 roughly: send cwnd bytes,
wait RTT for ACKS, then
send more bytes
last byte
available but cwnd
ACKed sent, but not-
not used
TCP rate ~
~ bytes/sec
yet ACKed RTT
(“in-flight”) last byte sent
 TCP sender limits transmission: LastByteSent- LastByteAcked < cwnd

 cwnd is dynamically adjusted in response to observed network
congestion (implementing TCP congestion control)
TCP Slow Start
Host A Host B
 when connection begins,
increase rate exponentially
one segm
until first loss event: ent
RTT
• initially cwnd = 1 MSS two segm
ents
• double cwnd every RTT
• done by incrementing cwnd
for every ACK received four segm
ents
 summary: initial rate is

slow, but ramps up
exponentially fast time

TCP Slow Start
 loss indicated by timeout:

• cwnd set to 1 MSS;
• window then grows exponentially (as in slow start) to threshold,
then grows linearly
• Set the threshold value ssthresh is equal to cwnd/2
 When the value of cwnd >= ssthresh, Slow Start ends and
Congestion Avoidance (CA) starts.
 loss indicated by 3 duplicate ACKs: TCP enters in the fast
recovery mode.
TCP: Congestion Avoidance (CA)
 Rather than doubling the cwnd value, cwnd is increased by just a
single MSS every RTT.
 TCP sender increase cwnd by MSS bytes (MSS/cwnd)
 When the congestion avoidance ends?
 Depends on the timeout events and triple duplicates
 dup ACKs indicate network capable of delivering some segments
 Fast Recovery: 3 dup ACKs
 TCP Tahoe always sets cwnd to 1 then grows exponentially (timeout or 3
duplicate acks) [Earlier Style]
 TCP Reno cut the cwnd in half window then grows linearly [New
Version]

TCP: from slow start to congestion avoidance
Q: when should the exponential
increase switch to linear?
X
A: when cwnd gets to 1/2 of its
value before timeout.
Implementation:
 variable ssthresh
 on loss event, ssthresh is set to
1/2 of cwnd just before loss event
Summary: TCP congestion control
New
New ACK!
.
ACK! new ACK
duplicate ACK
cwnd = cwnd + MSS (MSS/cwnd)
dupACKcount++ new ACK dupACKcount = 0
cwnd = cwnd+MSS transmit new segment(s), as allowed
dupACKcount = 0
L transmit new segment(s), as allowed
cwnd = 1 MSS
ssthresh = 64 KB cwnd > ssthresh
dupACKcount = 0
slow L congestion
start timeout avoidance
ssthresh = cwnd/2
cwnd = 1 MSS duplicate ACK
timeout dupACKcount = 0 dupACKcount++
ssthresh = cwnd/2 retransmit missing segment
cwnd = 1 MSS
dupACKcount = 0
retransmit missing segment
timeout
New
ACK!
ssthresh = cwnd/2
cwnd = 1 New ACK
dupACKcount = 0
cwnd = ssthresh dupACKcount == 3
dupACKcount == 3 retransmit missing segment dupACKcount = 0
ssthresh= cwnd/2 ssthresh= cwnd/2
cwnd = ssthresh + 3 cwnd = ssthresh + 3
retransmit missing segment
fast retransmit missing segment
recovery
duplicate ACK
cwnd = cwnd + MSS
transmit new segment(s), as allowed

Summary: TCP congestion control
 When CongWin is below Threshold, sender in slow start
phase, window grows exponentially.
 When CongWin is above Threshold, sender is in congestion

avoidance phase, window grows linearly.
 When a triple duplicate ACK occurs, Threshold set to

CongWin/2 and CongWin set to Threshold + 3.
 When timeout occurs, Threshold set to CongWin/2 and

CongWin is set to 1 MSS.

Summary: TCP Congestion
Control
 When CongWin is below Threshold, sender in slow start
phase, window grows exponentially.
 When CongWin is above Threshold, sender is in congestion

avoidance phase, window grows linearly.
 When a triple duplicate ACK occurs, Threshold set to

CongWin/2 and CongWin set to Threshold + 3.
 When timeout occurs, Threshold set to CongWin/2 and

CongWin is set to 1 MSS.
3-105
Summary: TCP Congestion
Control
TCP sender congestion control
3-106
Chapter 3: summary
 principles behind transport Up next:
layer services:  leaving the network
• multiplexing, demultiplexing “edge” (application,
• reliable data transfer transport layers)
• flow control  into the network “core”
• congestion control
 two network-layer
 instantiation, implementation chapters:
in the Internet • data plane
• UDP • control plane
• TCP

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Chapter 3

Transport Layer: 3-2

 provide logical communication mobile

between application processes national or global ISP

running on different hosts

 two transport protocols available to data link

Internet applications enterprise

Transport Layer: 3-4

 network layer: logical household analogy:

Transport Layer: 3-5

Transport Layer: 3-6

link to application via socket link

Transport Layer: 3-7

 TCP: Transmission Control Protocol mobile

 services not available:

• delay guarantees enterprise

transport network transport

transport network transport

Transport Layer: 3-11

Transport Layer: 3-12

transport network transport

Transport Layer: 3-13

transport network transport

Transport Layer: 3-14

application P1 P2 application socket

Transport Layer: 3-15

Transport Layer: 3-16

source port: 6428 source port: ?

source port: 9157 source port: ?

host: IP source IP,port: B,80 host: IP

Transport Layer: 3-21

Transport Layer: 3-24

Transport Layer: 3-25

application length, in bytes of

Transport Layer: 3-26

Transport Layer: 3-27

Transport Layer: 3-30

reliable service abstraction

Transport Layer: 3-33

sending receiving sending receiving

reliable service implementation

Transport Layer: 3-34

Transport Layer: 3-35

Transport Layer: 3-36

sender-side data receiver-side

udt_send() Header data Header data rdt_rcv()

Transport Layer: 3-38

 separate FSMs for sender, receiver:

Wait for rdt_send(data) Wait for rdt_rcv(packet)

Transport Layer: 3-39

How do humans recover from “errors” during conversation?

Transport Layer: 3-40

stop and wait

rdt_rcv(rcvpkt) && isACK(rcvpkt)

rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)

Transport Layer: 3-42

rdt_rcv(rcvpkt) && isACK(rcvpkt)

Note: “state” of receiver (did the receiver get my

Transport Layer: 3-43

rdt_rcv(rcvpkt) && isACK(rcvpkt)

rdt_rcv(rcvpkt) && notcorrupt(rcvpkt)

Transport Layer: 3-44

rdt_rcv(rcvpkt) && isACK(rcvpkt)